In currently used mobile radio telecommunication systems such as the existing GSM system, radio resources are limited due to the limited radio spectrum available and/or reserved for such purposes. The saving of radio resources is achieved by the usage of codecs (encoding-decoding devices) which operate at low bit rates. Namely, with such low-bit-rate codecs, the transmission rate on a radio interface is reduced to/expanded from 13 kbps in current GSM systems.
Furthermore, existing mobile radio telecommunication networks like GSM networks have to co-operate with existing public switched telephone networks PSTN such as the currently spreading ISDN network (Integrated Services Digital Network).
However, ISDN-PSTN networks primarily use a representation for speech data of 64 kbps in pulse code modulation (PCM), while the mobile radio telecommunication network uses the above mentioned 13 kbps representation.
Hence, interoperability at an interface I/F between the two types of networks has to be provided for, which is achieved by an interworking function unit also referred to as IWF unit or gateway unit (GW).
In case a call is forwarded from the ISDN-PSTN network to a mobile station MS as a radio terminal device of the radio telecommunication network (and vice versa), a speech coding functionality is required on both sides, i.e. on the network side and on the terminal device side. Namely, speech data of a call forwarded from the ISDN-PSTN network have to be coded in the network side for transmission via the radio interface (air interface Um), and if speech data of “an answer” are transmitted from the terminal device side, the speech data have likewise to be coded for transmission via the air interface.
Such a transmission of coded speech using time divisional multiplexing (TDM) between the radio access part or access network of the telecommunication network (e.g. a base transceiver station) and a speech codec part (e.g. transcoder and/or Transcoding Rate and Adaptation Unit TRAU) associated to, for example, a mobile Services Switching Center MSC as a part of a core network of the telecommunication network, according to GSM, is effected using so-called TRAU frames (for details, reference is made to GSM 08.60).
Now, if a call is established between two terminal devices MS_A and MS_B, speech data transmitted there between are normally transcoded twice. Namely, firstly speech is encoded in the terminal device MS_A and subsequently decoded in the network. Thus, the speech data are present in the 64 kbps PCM format. Thereafter, the speech is encoded again in the network for transmission to the terminal MS_B, where it is decoded upon being received.
Thus, the coding is performed twice, while such double coding adversely affects the quality of transmitted speech, which of course, is undesirable.
In order to prevent such coding being performed twice (also referred to as tandem coding), the European telecommunication Standards Institute (ETSI) has standardized a feature named Tandem Free Operation TFO, which is specified in detail in GSM 04.53. In short, it can be said that this TFO operation is based on an in-band signaling within the speech data stream of 64 kbps.
Herein above and up to now, a description has been made with only the mobile radio telecommunication network and a PSTN network such as the ISDN-PSTN being considered.
However, in most recent times also information networks like the Internet have widely spread and offer a number of increasingly used services. Such networks are based on a so-called Internet Protocol (IP). With such IP based networks, it is currently already possible to transmit speech data (either low bit-rate coded or PCM coded).
Nevertheless, the interoperability of such IP based networks with the currently existing radio telecommunication networks is rather limited and not optimal.
Interoperability between IP based networks and the currently existing (GSM based) radio telecommunication networks has been enhanced by using the above mentioned TFO method also in this respect.
However, although enhanced, this is not an optimum solution, since due to the adopted in-band signaling a connection (i.e. transmission channel) is rather slowly established and a minor speech degradation is still present.
Current developments are directed to so called third generation telecommunication systems, which are also referred to as UMTS (Universal Mobile Telecommunication System) systems.
However, while these UMTS systems are currently under development, there are no definite standards in terms of how the interoperability between those third generation telecommunication networks and IP based networks is to be assured and/or improved in comparison to existing telecommunication networks. Also, currently there are no specifications as regards the speech data transmission within a third generation network, i.e. speech transmission between a radio access network and a corresponding core network, which form together a UMTS network.